[Asterisk-Users] ISDN debugging and SIP dial-in issue]

[Marc SCHAEFER]

On Mon, Nov 17, 2003 at 02:08:30PM +0100, Philipp von Klitzing wrote:
> You should also add to sip.conf for [17476691152]:
> disallow=all
> allow=ulaw
> allow=alaw

This was the key. I now hear the voice prompts correctly.

> rather use the "new" syntax for the Dial application like 
> "Dial(Modem/g1/012345,20,rt)".

I now use the CAPI driver, which works fine, including
call transfer by copying.

My dial plan:

; Dial plan:
; 0 001 747 xxx xxxx   routed to sipphone.com with 1152's registration
; 0 001 800 xxx xxxx   routed to sipphone.com with 1152's registration
; 0 xxxxxxxxx*         routed to ISDN dialup (allowed from local SIP only)
; 100                  the demo
; NOTES
;    - Calls from sipphone.com go to default, which is xfertomarclocal (SIP)
;    - isdn-dial up can also dial to sipphone.com or go to the demo through
;      an escape.

My configurations:

   - dial in from ISDN goes first to SIP phone then to external
     mobile phone through ISDN call copying:
     (could also be done through not answering and deflecting)

[isdn-in]
include => no-suckers
include => xfertomarc

   - local SIP phone dials out with '0' on ISDN. Special cases go to
     sipphone.com

[local-sip-in]
include => sipphone.com
include => isdn-dial-out ; order matters.
exten => 100,1,Goto(demo,s,1)

[xfertomarc]
include => xfertomarclocal
exten => s,3,Goto(xfertomobile,s,4) ; ---

[xfertomarclocal]
exten => s,1,Answer  ; Probably could also defer answering
exten => s,2,Dial(sip/17476691152,30)       ; schaefer

[xfertomobile]
exten => s,1,Wait,1                     ; Wait a second, just for fun
exten => s,2,Answer                     ; Answer the line
;exten => s,3,Playback(transfer,skip)    ; schaefer
exten => s,3,Background(transfer)    ; schaefer
exten => s,4,Dial,CAPI/8414014:xxxxxxxxx|30|r
exten => s,5,Playback(publicar-extbusy,skip) ; schaefer
exten => s,6,Hangup                     ; schaefer

[sipphone.com]
exten => s,1,Answer ; dummy for Goto
exten => s,2,DigitTimeout,10
exten => s,3,ResponseTimeout,10
exten => i,1,Playback(invalid)          ; "That's not valid, try again"
exten => _001747NXXXXXX,1,SetCallerID(${CALLERIDNUM})
exten => _001747NXXXXXX,2,SetCIDName(${CALLERIDNUM})
exten => _001747NXXXXXX,3,Dial(Sip/${EXTEN:2}@sipphone)
exten => _001747NXXXXXX,4,Playback(invalid)
;exten => _001747NXXXXXX,5,Hangup

exten => _001800NXXXXXX,1,SetCallerID(${CALLERIDNUM})
exten => _001800NXXXXXX,2,SetCIDName(${CALLERIDNUM})
exten => _001800NXXXXXX,3,Dial(Sip/${EXTEN:2}@sipphone)
exten => _001800NXXXXXX,4,Playback(invalid)
;exten => _001800NXXXXXX,5,Hangup

Thank you for your help.

I still have one -- probably NAT related -- issue with
contacting SIPphone.com (no sound). Will document it.